mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-20 00:57:49 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/call
History
Danil Chapovalov 3c5850148e Allow to create webrtc::Call with Environment 
instead of passing utilities one by one

Bug: webrtc:15656
Change-Id: I1f3bf7ae66dcc62bbf17d81c927aabe748b42163
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328540
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41256}
2023-11-28 10:26:56 +00:00
..
adaptation Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
test Revert "Clean up last_packet_received_time_ as it's no longer used." 2023-09-25 08:49:53 +00:00
audio_receive_stream.cc Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
audio_receive_stream.h Propagate time of the last received packet with Timestamp type 2023-06-02 14:29:19 +00:00
audio_send_stream.cc Reland "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-06 14:26:55 +00:00
audio_send_stream.h Replace "rcvd" with "received" for readability 2023-04-24 15:30:07 +00:00
audio_sender.h Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api 2020-01-13 18:31:30 +00:00
audio_state.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_state.h Async audio processing API 2020-10-02 12:33:34 +00:00
bitrate_allocator.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 2022-03-09 13:23:21 +00:00
bitrate_allocator.h Use backticks not vertical bars to denote variables in comments for /call 2021-07-27 18:29:33 +00:00
bitrate_allocator_unittest.cc Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
bitrate_estimator_tests.cc Reland: FrameGeneratorCapturer: don't generate video before Start is called 2023-06-29 14:47:05 +00:00
BUILD.gn Allow to create webrtc::Call with Environment 2023-11-28 10:26:56 +00:00
call.cc Allow to create webrtc::Call with Environment 2023-11-28 10:26:56 +00:00
call.h Cleanup Call construction 2023-10-16 06:34:26 +00:00
call_config.cc Allow to create webrtc::Call with Environment 2023-11-28 10:26:56 +00:00
call_config.h Allow to create webrtc::Call with Environment 2023-11-28 10:26:56 +00:00
call_factory.cc Cleanup Call construction 2023-10-16 06:34:26 +00:00
call_factory.h Cleanup Call construction 2023-10-16 06:34:26 +00:00
call_perf_tests.cc Cleanup Call construction 2023-10-16 06:34:26 +00:00
call_unittest.cc Allow to create webrtc::Call with Environment 2023-11-28 10:26:56 +00:00
degraded_call.cc Remove internal overrides using old SendRtp and SendRtcp interfaces. 2023-08-15 13:20:21 +00:00
degraded_call.h Remove internal overrides using old SendRtp and SendRtcp interfaces. 2023-08-15 13:20:21 +00:00
DEPS SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. 2021-08-30 10:20:55 +00:00
fake_network_pipe.cc Delete unused constructor of FakeNetworkPipe 2023-08-18 13:07:10 +00:00
fake_network_pipe.h Delete unused constructor of FakeNetworkPipe 2023-08-18 13:07:10 +00:00
fake_network_pipe_unittest.cc Reland "Delete PacketReceiver::DeliverPacket from all implementations" 2023-01-25 18:18:29 +00:00
flexfec_receive_stream.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h stats: implement flexfec fecBytesReceived stats for FlexFEC 2023-06-21 13:04:31 +00:00
flexfec_receive_stream_impl.cc stats: implement flexfec fecBytesReceived stats for FlexFEC 2023-06-21 13:04:31 +00:00
flexfec_receive_stream_impl.h stats: implement flexfec fecBytesReceived stats for FlexFEC 2023-06-21 13:04:31 +00:00
flexfec_receive_stream_unittest.cc Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
OWNERS Update OWNERS for call/ 2022-06-03 12:01:46 +00:00
packet_receiver.h Allow injecting packets of type Any to Call::DeliverRtpPacket 2023-03-29 06:36:17 +00:00
rampup_tests.cc Replace WebRTC-QuickPerfTest field trial with a flag 2023-10-10 08:59:10 +00:00
rampup_tests.h Stop overriding extensions in rampup tests 2023-01-25 13:18:49 +00:00
receive_stream.h Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
receive_time_calculator.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
receive_time_calculator.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
receive_time_calculator_unittest.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2 2022-03-09 22:17:52 +00:00
rtp_bitrate_configurator.cc Allow setting a bandwidth cap for relayed connections. 2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h Remove RTC_DISALLOW_COPY_AND_ASSIGN more. 2022-01-20 11:00:18 +00:00
rtp_bitrate_configurator_unittest.cc Revert "In RtpBitrateConfigurator ignore new parameters when set to default values." 2020-01-10 16:39:51 +00:00
rtp_config.cc Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED 2021-11-15 21:44:59 +00:00
rtp_config.h Introduce support for video packet batching. 2023-05-08 16:24:03 +00:00
rtp_demuxer.cc Remove SSRCs from libSRTP when removing them from the rtp_demuxer 2023-11-08 10:24:10 +00:00
rtp_demuxer.h Remove SSRCs from libSRTP when removing them from the rtp_demuxer 2023-11-08 10:24:10 +00:00
rtp_demuxer_unittest.cc Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
rtp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_payload_params.cc Add codec name H265 to support H265 in WebRTC 2023-09-20 09:25:32 +00:00
rtp_payload_params.h For VP9 assume max number of spatial layers to simulate generic descriptor 2022-06-08 11:36:54 +00:00
rtp_payload_params_unittest.cc When simulating chains from VP9 codec specific info support first_active_layer > 0 2023-08-03 13:19:00 +00:00
rtp_stream_receiver_controller.cc Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived 2022-12-22 14:04:21 +00:00
rtp_stream_receiver_controller.h Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived 2022-12-22 14:04:21 +00:00
rtp_stream_receiver_controller_interface.h Demote RtpStreamReceiverController AddSink/RemoveSink to private 2022-07-06 09:31:54 +00:00
rtp_transport_config.h Per default set PacingController burst interval to 40ms 2023-11-28 07:53:50 +00:00
rtp_transport_controller_send.cc Per default set PacingController burst interval to 40ms 2023-11-28 07:53:50 +00:00
rtp_transport_controller_send.h Revert "Clean up last_packet_received_time_ as it's no longer used." 2023-09-25 08:49:53 +00:00
rtp_transport_controller_send_factory.h Refactor some config plumbing in call/. 2022-11-16 09:18:40 +00:00
rtp_transport_controller_send_factory_interface.h Delete Call dependency on ProcessThread as unused 2022-06-21 08:59:38 +00:00
rtp_transport_controller_send_interface.h Revert "Clean up last_packet_received_time_ as it's no longer used." 2023-09-25 08:49:53 +00:00
rtp_video_sender.cc Make field trial string DisableRtxRateLimiter enabled by default. 2023-10-27 12:33:58 +00:00
rtp_video_sender.h Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController. 2023-06-01 07:51:56 +00:00
rtp_video_sender_interface.h Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" 2022-12-02 12:03:25 +00:00
rtp_video_sender_unittest.cc Per default set PacingController burst interval to 40ms 2023-11-28 07:53:50 +00:00
rtx_receive_stream.cc Updated associated payload types without recreating receive streams. 2022-08-16 13:38:24 +00:00
rtx_receive_stream.h Updated associated payload types without recreating receive streams. 2022-08-16 13:38:24 +00:00
rtx_receive_stream_unittest.cc Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +00:00
simulated_network.cc Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." 2022-11-06 13:14:26 +00:00
simulated_network.h Export webrtc::SimulatedNetwork for Chrome component builds 2023-11-27 16:03:23 +00:00
simulated_network_unittest.cc Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." 2022-11-06 13:14:26 +00:00
simulated_packet_receiver.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
version.cc Update WebRTC code version (2023-11-27T04:13:30). 2023-11-27 05:49:44 +00:00
version.h Add WebRTC code freshness version string. 2020-12-14 16:22:35 +00:00
video_receive_stream.cc Add missing comma in VideoReceiveStreamInterface::Stats::ToString 2023-10-17 10:42:06 +00:00
video_receive_stream.h Remove default "unknown" encoderImplementation/decoderImplementation 2023-06-22 11:49:58 +00:00
video_send_stream.cc Cleanup usasge of ReportBlockData::report_block accessor 2023-05-05 09:56:30 +00:00
video_send_stream.h VideoStreamEncoder: Clean up drop handling and update rects. 2023-11-23 17:19:33 +00:00